AG_GST_CHECK_PLUGIN(rawparse)
AG_GST_CHECK_PLUGIN(removesilence)
AG_GST_CHECK_PLUGIN(rist)
+AG_GST_CHECK_PLUGIN(rtp)
AG_GST_CHECK_PLUGIN(sdp)
AG_GST_CHECK_PLUGIN(segmentclip)
AG_GST_CHECK_PLUGIN(siren)
gst/rawparse/Makefile
gst/removesilence/Makefile
gst/rist/Makefile
+gst/rtp/Makefile
gst/sdp/Makefile
gst/segmentclip/Makefile
gst/siren/Makefile
'ivfparse', 'ivtc', 'jp2kdecimator', 'jpegformat', 'librfb',
'midi', 'mpegdemux', 'mpegpsmux', 'mpegtsdemux', 'mpegtsmux',
'mxf', 'netsim', 'onvif', 'pcapparse', 'pnm', 'proxy',
- 'rawparse', 'removesilence', 'rist', 'sdp', 'segmentclip',
+ 'rawparse', 'removesilence', 'rist', 'rtp', 'sdp', 'segmentclip',
'siren', 'smooth', 'speed', 'subenc', 'timecode',
'videofilters', 'videoframe_audiolevel', 'videoparsers',
'videosignal', 'vmnc', 'y4m', 'yadif']
--- /dev/null
+plugin_LTLIBRARIES = libgstrtpmanagerbad.la
+
+libgstrtpmanagerbad_la_SOURCES = \
+ gstrtp-utils.c \
+ gstrtpsink.c \
+ gstrtpsrc.c \
+ plugin.c
+
+libgstrtpmanagerbad_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+libgstrtpmanagerbad_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS)
+libgstrtpmanagerbad_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = \
+ gstrtp-utils.h \
+ gstrtpcaps.h \
+ gstrtpsink.h \
+ gstrtpsrc.h
--- /dev/null
+/*
+ * See: https://bugzilla.gnome.org/show_bug.cgi?id=779765
+ */
+
+#include "gstrtp-utils.h"
+
+static void
+gst_rtp_utils_uri_query_foreach (const gchar * key, const gchar * value,
+ GObject * src)
+{
+ if (key == NULL) {
+ GST_WARNING_OBJECT (src, "Refusing to use empty key.");
+ return;
+ }
+
+ if (value == NULL) {
+ GST_WARNING_OBJECT (src, "Refusing to use NULL for key %s.", key);
+ return;
+ }
+
+ GST_DEBUG_OBJECT (src, "Setting property '%s' to '%s'", key, value);
+ gst_util_set_object_arg (src, key, value);
+}
+
+void
+gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri)
+{
+ GHashTable *hash_table;
+
+ g_return_if_fail (uri != NULL);
+ hash_table = gst_uri_get_query_table (uri);
+
+ if (hash_table) {
+ g_hash_table_foreach (hash_table,
+ (GHFunc) gst_rtp_utils_uri_query_foreach, obj);
+
+ g_hash_table_unref (hash_table);
+ }
+}
--- /dev/null
+#ifndef __GST_RTP_UTILS_H__
+#define __GST_RTP_UTILS_H__
+
+#include <gst/gst.h>
+
+void gst_rtp_utils_set_properties_from_uri_query (GObject * obj, const GstUri * uri);
+
+#endif
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION: gstrtsinkp
+ * @title: GstRtpSink
+ * @short description: element with Uri interface to stream RTP data to
+ * the network.
+ *
+ * RTP (RFC 3550) is a protocol to stream media over the network while
+ * retaining the timing information and providing enough information to
+ * reconstruct the correct timing domain by the receiver.
+ *
+ * The RTP data port should be even, while the RTCP port should be
+ * odd. The URI that is entered defines the data port, the RTCP port will
+ * be allocated to the next port.
+ *
+ * This element hooks up the correct sockets to support both RTP as the
+ * accompanying RTCP layer.
+ *
+ * This Bin handles streaming RTP payloaded data on the network.
+ *
+ * This element also implements the URI scheme `rtp://` allowing to send
+ * data on the network by bins that allow use the URI to determine the sink.
+ * The RTP URI handler also allows setting properties through the URI query.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gio/gio.h>
+
+#include "gstrtpsink.h"
+#include "gstrtp-utils.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
+#define GST_CAT_DEFAULT gst_rtp_sink_debug
+
+#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
+#define DEFAULT_PROP_TTL 64
+#define DEFAULT_PROP_TTL_MC 1
+
+enum
+{
+ PROP_0,
+
+ PROP_URI,
+ PROP_TTL,
+ PROP_TTL_MC,
+
+ PROP_LAST
+};
+
+static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
+ gpointer iface_data);
+
+#define gst_rtp_sink_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
+ GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
+
+#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
+#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
+#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
+ GST_PAD_SINK,
+ GST_PAD_REQUEST,
+ GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStateChangeReturn
+gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
+
+static void
+gst_rtp_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRtpSink *self = GST_RTP_SINK (object);
+
+ switch (prop_id) {
+ case PROP_URI:{
+ GstUri *uri = NULL;
+
+ GST_RTP_SINK_LOCK (object);
+ uri = gst_uri_from_string (g_value_get_string (value));
+ if (uri == NULL)
+ break;
+
+ if (self->uri)
+ gst_uri_unref (self->uri);
+ self->uri = uri;
+ /* RTP data ports should be even according to RFC 3550, while the
+ * RTCP is sent on odd ports. Just warn if there is a mismatch. */
+ if (gst_uri_get_port (self->uri) % 2)
+ GST_WARNING_OBJECT (self,
+ "Port %u is not even, this is not standard (see RFC 3550).",
+ gst_uri_get_port (self->uri));
+
+ gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
+ GST_RTP_SINK_UNLOCK (object);
+ break;
+ }
+ case PROP_TTL:
+ self->ttl = g_value_get_int (value);
+ break;
+ case PROP_TTL_MC:
+ self->ttl_mc = g_value_get_int (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRtpSink *self = GST_RTP_SINK (object);
+
+ switch (prop_id) {
+ case PROP_URI:
+ GST_RTP_SINK_LOCK (object);
+ if (self->uri)
+ g_value_take_string (value, gst_uri_to_string (self->uri));
+ else
+ g_value_set_string (value, NULL);
+ GST_RTP_SINK_UNLOCK (object);
+ break;
+ case PROP_TTL:
+ g_value_set_int (value, self->ttl);
+ break;
+ case PROP_TTL_MC:
+ g_value_set_int (value, self->ttl_mc);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_sink_finalize (GObject * gobject)
+{
+ GstRtpSink *self = GST_RTP_SINK (gobject);
+
+ if (self->uri)
+ gst_uri_unref (self->uri);
+
+ g_mutex_clear (&self->lock);
+ G_OBJECT_CLASS (parent_class)->finalize (gobject);
+}
+
+static gboolean
+gst_rtp_sink_setup_elements (GstRtpSink * self)
+{
+ /*GstPad *pad; */
+ GSocket *socket;
+ GInetAddress *addr;
+ gchar name[48];
+ GstCaps *caps;
+
+ /* Should not be NULL */
+ g_return_val_if_fail (self->uri != NULL, FALSE);
+
+ /* if not already configured */
+ if (self->funnel_rtp == NULL) {
+ self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
+ if (self->funnel_rtp == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "funnel_rtp element is not available"));
+ return FALSE;
+ }
+
+ self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
+ if (self->funnel_rtcp == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "funnel_rtcp element is not available"));
+ return FALSE;
+ }
+
+ self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
+ if (self->rtp_sink == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtp_sink element is not available"));
+ return FALSE;
+ }
+
+ self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
+ if (self->rtcp_src == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtcp_src element is not available"));
+ return FALSE;
+ }
+
+ self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
+ if (self->rtcp_sink == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtcp_sink element is not available"));
+ return FALSE;
+ }
+
+ gst_bin_add (GST_BIN (self), self->funnel_rtp);
+ gst_bin_add (GST_BIN (self), self->funnel_rtcp);
+
+ /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
+ * not all at the same moment */
+ g_object_set (self->rtp_sink,
+ "host", gst_uri_get_host (self->uri),
+ "port", gst_uri_get_port (self->uri),
+ "ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
+
+ gst_bin_add (GST_BIN (self), self->rtp_sink);
+
+ g_object_set (self->rtcp_sink,
+ "host", gst_uri_get_host (self->uri),
+ "port", gst_uri_get_port (self->uri) + 1,
+ "ttl", self->ttl, "ttl-mc", self->ttl_mc,
+ /* Set false since we're reusing a socket */
+ "auto-multicast", FALSE, NULL);
+
+ gst_bin_add (GST_BIN (self), self->rtcp_sink);
+
+ /* no need to set address if unicast */
+ caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ g_object_set (self->rtcp_src,
+ "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
+ if (g_inet_address_get_is_multicast (addr)) {
+ g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
+ NULL);
+ }
+ g_object_unref (addr);
+
+ gst_bin_add (GST_BIN (self), self->rtcp_src);
+
+ gst_element_link (self->funnel_rtp, self->rtp_sink);
+ gst_element_link (self->funnel_rtcp, self->rtcp_sink);
+
+ gst_element_sync_state_with_parent (self->funnel_rtp);
+ gst_element_sync_state_with_parent (self->funnel_rtcp);
+ gst_element_sync_state_with_parent (self->rtp_sink);
+ gst_element_sync_state_with_parent (self->rtcp_src);
+
+ g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
+ g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
+
+ gst_element_sync_state_with_parent (self->rtcp_sink);
+
+ }
+
+ /* pads are all named */
+ g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
+
+ g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
+
+ g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
+
+ return TRUE;
+}
+
+static GstPad *
+gst_rtp_sink_request_new_pad (GstElement * element,
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
+{
+ GstRtpSink *self = GST_RTP_SINK (element);
+ GstPad *pad = NULL;
+
+ if (self->rtpbin == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtpbin element is not available"));
+ return NULL;
+ }
+
+ if (gst_rtp_sink_setup_elements (self) == FALSE)
+ return NULL;
+
+ GST_RTP_SINK_LOCK (self);
+
+ pad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_%u");
+ g_return_val_if_fail (pad != NULL, NULL);
+
+ GST_RTP_SINK_UNLOCK (self);
+
+ return pad;
+}
+
+static void
+gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
+{
+ GstRtpSink *self = GST_RTP_SINK (element);
+ GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+
+ GST_RTP_SINK_LOCK (self);
+ gst_element_release_request_pad (self->rtpbin, rpad);
+ gst_object_unref (rpad);
+
+ gst_pad_set_active (pad, FALSE);
+ gst_element_remove_pad (GST_ELEMENT (self), pad);
+
+ GST_RTP_SINK_UNLOCK (self);
+}
+
+static void
+gst_rtp_sink_class_init (GstRtpSinkClass * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+ gobject_class->set_property = gst_rtp_sink_set_property;
+ gobject_class->get_property = gst_rtp_sink_get_property;
+ gobject_class->finalize = gst_rtp_sink_finalize;
+ gstelement_class->change_state = gst_rtp_sink_change_state;
+
+ gstelement_class->request_new_pad =
+ GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
+ gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
+
+ /**
+ * GstRtpSink:uri:
+ *
+ * uri to stream RTP to. All GStreamer parameters can be
+ * encoded in the URI, this URI format is RFC compliant.
+ */
+ g_object_class_install_property (gobject_class, PROP_URI,
+ g_param_spec_string ("uri", "URI",
+ "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSink:ttl:
+ *
+ * Set the unicast TTL parameter.
+ */
+ g_object_class_install_property (gobject_class, PROP_TTL,
+ g_param_spec_int ("ttl", "Unicast TTL",
+ "Used for setting the unicast TTL parameter",
+ 0, 255, DEFAULT_PROP_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSink:ttl-mc:
+ *
+ * Set the multicast TTL parameter.
+ */
+ g_object_class_install_property (gobject_class, PROP_TTL_MC,
+ g_param_spec_int ("ttl-mc", "Multicast TTL",
+ "Used for setting the multicast TTL parameter", 0, 255,
+ DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&sink_template));
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP Sink element",
+ "Generic/Bin/Sink",
+ "Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
+}
+
+static void
+gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
+ GstElement * new_element, gpointer data)
+{
+ GstRtpSink *self = GST_RTP_SINK (data);
+ GST_INFO_OBJECT (self,
+ "Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
+ new_element);
+}
+
+static void
+gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
+ gpointer data)
+{
+ GstRtpSink *self = GST_RTP_SINK (data);
+ GstCaps *caps = gst_pad_query_caps (pad, NULL);
+ GstPad *upad;
+
+ /* Expose RTP data pad only */
+ GST_INFO_OBJECT (self,
+ "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
+ GST_PTR_FORMAT ".", element, pad, caps);
+
+ /* Sanity checks */
+ if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
+ /* Src pad, do not expose */
+ gst_caps_unref (caps);
+ return;
+ }
+
+ if (G_LIKELY (caps)) {
+ GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
+
+ if (gst_caps_can_intersect (caps, ref_caps)) {
+ /* SRC RTCP caps, do not expose */
+ gst_caps_unref (ref_caps);
+ gst_caps_unref (caps);
+
+ return;
+ }
+ gst_caps_unref (ref_caps);
+ } else {
+ GST_ERROR_OBJECT (self, "Pad with no caps detected.");
+ gst_caps_unref (caps);
+
+ return;
+ }
+ gst_caps_unref (caps);
+
+ upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
+ if (upad == NULL) {
+ GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
+ gst_caps_unref (caps);
+
+ return;
+ }
+ GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
+ gst_pad_link (pad, upad);
+ gst_object_unref (upad);
+}
+
+static void
+gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
+ gpointer data)
+{
+ GstRtpSink *self = GST_RTP_SINK (data);
+ GST_INFO_OBJECT (self,
+ "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
+ pad);
+}
+
+static gboolean
+gst_rtp_sink_setup_rtpbin (GstRtpSink * self)
+{
+ self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
+ if (self->rtpbin == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtpbin element is not available"));
+ return FALSE;
+ }
+
+ /* Add rtpbin callbacks to monitor the operation of rtpbin */
+ g_signal_connect (self->rtpbin, "element-added",
+ G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
+ g_signal_connect (self->rtpbin, "pad-added",
+ G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
+ g_signal_connect (self->rtpbin, "pad-removed",
+ G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
+
+ gst_bin_add (GST_BIN (self), self->rtpbin);
+
+ gst_element_sync_state_with_parent (self->rtpbin);
+
+ return TRUE;
+}
+
+static GstStateChangeReturn
+gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
+{
+ GstRtpSink *self = GST_RTP_SINK (element);
+ GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+ GST_DEBUG_OBJECT (self, "changing state: %s => %s",
+ gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
+ gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ return ret;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+
+static void
+gst_rtp_sink_init (GstRtpSink * self)
+{
+ self->rtpbin = NULL;
+ self->funnel_rtp = NULL;
+ self->funnel_rtcp = NULL;
+ self->rtp_sink = NULL;
+ self->rtcp_src = NULL;
+ self->rtcp_sink = NULL;
+
+ self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
+ self->ttl = DEFAULT_PROP_TTL;
+ self->ttl_mc = DEFAULT_PROP_TTL_MC;
+
+ if (gst_rtp_sink_setup_rtpbin (self) == FALSE)
+ return;
+
+ GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
+ gst_bin_set_suppressed_flags (GST_BIN (self),
+ GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
+
+ g_mutex_init (&self->lock);
+}
+
+static guint
+gst_rtp_sink_uri_get_type (GType type)
+{
+ return GST_URI_SINK;
+}
+
+static const gchar *const *
+gst_rtp_sink_uri_get_protocols (GType type)
+{
+ static const gchar *protocols[] = { (char *) "rtp", NULL };
+
+ return protocols;
+}
+
+static gchar *
+gst_rtp_sink_uri_get_uri (GstURIHandler * handler)
+{
+ GstRtpSink *self = (GstRtpSink *) handler;
+
+ return gst_uri_to_string (self->uri);
+}
+
+static gboolean
+gst_rtp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+ GError ** error)
+{
+ GstRtpSink *self = (GstRtpSink *) handler;
+
+ g_object_set (G_OBJECT (self), "uri", uri, NULL);
+
+ return TRUE;
+}
+
+static void
+gst_rtp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+ GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+ iface->get_type = gst_rtp_sink_uri_get_type;
+ iface->get_protocols = gst_rtp_sink_uri_get_protocols;
+ iface->get_uri = gst_rtp_sink_uri_get_uri;
+ iface->set_uri = gst_rtp_sink_uri_set_uri;
+}
+
+/* ex: set tabstop=2 shiftwidth=2 expandtab: */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_SINK_H__
+#define __GST_RTP_SINK_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_RTP_SINK \
+ (gst_rtp_sink_get_type())
+#define GST_RTP_SINK(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SINK, GstRtpSink))
+#define GST_RTP_SINK_CAST(obj) \
+ ((GstRtpSink *) obj)
+#define GST_RTP_SINK_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SINK, GstRtpSinkClass))
+#define GST_IS_RTP_SINK(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SINK))
+#define GST_IS_RTP_SINK_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SINK))
+
+typedef struct _GstRtpSink GstRtpSink;
+typedef struct _GstRtpSinkClass GstRtpSinkClass;
+
+struct _GstRtpSink
+{
+ GstBin parent;
+
+ GstBin parent_instance;
+
+ /* Properties */
+ GstUri *uri;
+ gint ttl;
+ gint ttl_mc;
+
+ /* Internal elements */
+ GstElement *rtpbin;
+ GstElement *funnel_rtp;
+ GstElement *funnel_rtcp;
+ GstElement *rtp_sink;
+ GstElement *rtcp_src;
+ GstElement *rtcp_sink;
+
+ GMutex lock;
+};
+
+struct _GstRtpSinkClass
+{
+ GstBinClass parent;
+};
+
+GType gst_rtp_sink_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_RTP_SINK_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION: gstrtpsrc
+ * @title: GstRtpSrc
+ * @short description: element with Uri interface to get RTP data from
+ * the network.
+ *
+ * RTP (RFC 3550) is a protocol to stream media over the network while
+ * retaining the timing information and providing enough information to
+ * reconstruct the correct timing domain by the receiver.
+ *
+ * The RTP data port should be even, while the RTCP port should be
+ * odd. The URI that is entered defines the data port, the RTCP port will
+ * be allocated to the next port.
+ *
+ * This element hooks up the correct sockets to support both RTP as the
+ * accompanying RTCP layer.
+ *
+ * This Bin handles taking in of data from the network and provides the
+ * RTP payloaded data.
+ *
+ * This element also implements the URI scheme `rtp://` allowing to render
+ * RTP streams in GStreamer based media players. The RTP URI handler also
+ * allows setting properties through the URI query.
+ */
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/net/net.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#include "gstrtpsrc.h"
+#include "gstrtp-utils.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
+#define GST_CAT_DEFAULT gst_rtp_src_debug
+
+#define DEFAULT_PROP_TTL 64
+#define DEFAULT_PROP_TTL_MC 1
+#define DEFAULT_PROP_ENCODING_NAME NULL
+#define DEFAULT_PROP_LATENCY 200
+
+#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
+
+enum
+{
+ PROP_0,
+
+ PROP_URI,
+ PROP_TTL,
+ PROP_TTL_MC,
+ PROP_ENCODING_NAME,
+ PROP_LATENCY,
+
+ PROP_LAST
+};
+
+static void gst_rtp_src_uri_handler_init (gpointer g_iface,
+ gpointer iface_data);
+
+#define gst_rtp_src_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init);
+ GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source"));
+
+#define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock)
+#define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj)))
+#define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj)))
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
+ GST_PAD_SRC,
+ GST_PAD_SOMETIMES,
+ GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStateChangeReturn
+gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
+
+/**
+ * gst_rtp_src_rtpbin_erquest_pt_map_cb:
+ * @self: The current #GstRtpSrc object
+ *
+ * #GstRtpBin callback to map a pt on RTP caps.
+ *
+ * Returns: (transfer none): the guess on the RTP caps based on the PT
+ * and caps.
+ */
+static GstCaps *
+gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id,
+ guint pt, gpointer data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (data);
+ const GstRTPPayloadInfo *p = NULL;
+
+ GST_DEBUG_OBJECT (self,
+ "Requesting caps for session-id 0x%x and pt %u.", session_id, pt);
+
+ /* the encoding-name has more relevant information */
+ if (self->encoding_name != NULL) {
+ /* Unfortunately, the media needs to be passed in the function. Since
+ * it is not known, try for video if video not found. */
+ p = gst_rtp_payload_info_for_name ("video", self->encoding_name);
+ if (p == NULL)
+ p = gst_rtp_payload_info_for_name ("audio", self->encoding_name);
+
+ }
+
+ /* Static payload types, this is a simple lookup */
+ if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
+ p = gst_rtp_payload_info_for_pt (pt);
+ }
+
+ if (p != NULL) {
+ GstCaps *ret = gst_caps_new_simple ("application/x-rtp",
+ "encoding-name", G_TYPE_STRING, p->encoding_name,
+ "clock-rate", G_TYPE_INT, p->clock_rate,
+ "media", G_TYPE_STRING, p->media, NULL);
+
+ GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret);
+
+ return ret;
+ }
+
+ GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and"
+ " the encoding-name was not set.");
+ return NULL;
+}
+
+static void
+gst_rtp_src_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRtpSrc *self = GST_RTP_SRC (object);
+ GstCaps *caps;
+
+ switch (prop_id) {
+ case PROP_URI:{
+ GstUri *uri = NULL;
+
+ GST_RTP_SRC_LOCK (object);
+ uri = gst_uri_from_string (g_value_get_string (value));
+ if (uri == NULL)
+ break;
+
+ if (self->uri)
+ gst_uri_unref (self->uri);
+ self->uri = uri;
+ if (gst_uri_get_port (self->uri) % 2)
+ GST_WARNING_OBJECT (self,
+ "Port %u is not even, this is not standard (see RFC 3550).",
+ gst_uri_get_port (self->uri));
+ gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
+ GST_RTP_SRC_UNLOCK (object);
+ break;
+ }
+ case PROP_TTL:
+ self->ttl = g_value_get_int (value);
+ break;
+ case PROP_TTL_MC:
+ self->ttl_mc = g_value_get_int (value);
+ break;
+ case PROP_ENCODING_NAME:
+ g_free (self->encoding_name);
+ self->encoding_name = g_value_dup_string (value);
+ if (self->rtp_src) {
+ caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self);
+ g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL);
+ gst_caps_unref (caps);
+ }
+ break;
+ case PROP_LATENCY:
+ self->latency = g_value_get_uint (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_src_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRtpSrc *self = GST_RTP_SRC (object);
+
+ switch (prop_id) {
+ case PROP_URI:
+ GST_RTP_SRC_LOCK (object);
+ if (self->uri)
+ g_value_take_string (value, gst_uri_to_string (self->uri));
+ else
+ g_value_set_string (value, NULL);
+ GST_RTP_SRC_UNLOCK (object);
+ break;
+ case PROP_TTL:
+ g_value_set_int (value, self->ttl);
+ break;
+ case PROP_TTL_MC:
+ g_value_set_int (value, self->ttl_mc);
+ break;
+ case PROP_ENCODING_NAME:
+ g_value_set_string (value, self->encoding_name);
+ break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, self->latency);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_src_finalize (GObject * gobject)
+{
+ GstRtpSrc *self = GST_RTP_SRC (gobject);
+
+ if (self->uri)
+ gst_uri_unref (self->uri);
+ g_free (self->encoding_name);
+
+ g_mutex_clear (&self->lock);
+ G_OBJECT_CLASS (parent_class)->finalize (gobject);
+}
+
+static void
+gst_rtp_src_class_init (GstRtpSrcClass * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+ gobject_class->set_property = gst_rtp_src_set_property;
+ gobject_class->get_property = gst_rtp_src_get_property;
+ gobject_class->finalize = gst_rtp_src_finalize;
+ gstelement_class->change_state = gst_rtp_src_change_state;
+
+ /**
+ * GstRtpSrc:uri:
+ *
+ * uri to an RTP from. All GStreamer parameters can be
+ * encoded in the URI, this URI format is RFC compliant.
+ */
+ g_object_class_install_property (gobject_class, PROP_URI,
+ g_param_spec_string ("uri", "URI",
+ "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSrc:ttl:
+ *
+ * Set the unicast TTL parameter. In RTP this of importance for RTCP.
+ */
+ g_object_class_install_property (gobject_class, PROP_TTL,
+ g_param_spec_int ("ttl", "Unicast TTL",
+ "Used for setting the unicast TTL parameter",
+ 0, 255, DEFAULT_PROP_TTL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSrc:ttl-mc:
+ *
+ * Set the multicast TTL parameter. In RTP this of importance for RTCP.
+ */
+ g_object_class_install_property (gobject_class, PROP_TTL_MC,
+ g_param_spec_int ("ttl-mc", "Multicast TTL",
+ "Used for setting the multicast TTL parameter", 0, 255,
+ DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSrc:encoding-name:
+ *
+ * Set the encoding name of the stream to use. This is a short-hand for
+ * the full caps and maps typically to the encoding-name in the RTP caps.
+ */
+ g_object_class_install_property (gobject_class, PROP_ENCODING_NAME,
+ g_param_spec_string ("encoding-name", "Caps encoding name",
+ "Encoding name use to determine caps parameters",
+ DEFAULT_PROP_ENCODING_NAME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpSrc:latency:
+ *
+ * Set the size of the latency buffer in the
+ * GstRtpBin/GstRtpJitterBuffer to compensate for network jitter.
+ */
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Buffer latency in ms",
+ "Default amount of ms to buffer in the jitterbuffers", 0,
+ G_MAXUINT, DEFAULT_PROP_LATENCY,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&src_template));
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP Source element",
+ "Generic/Bin/Src",
+ "Simple RTP src", "Marc Leeman <marc.leeman@gmail.com>");
+}
+
+static void
+gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
+ gpointer data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (data);
+ GstCaps *caps = gst_pad_query_caps (pad, NULL);
+ GstPad *upad;
+ gchar name[48];
+
+ /* Expose RTP data pad only */
+ GST_INFO_OBJECT (self,
+ "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
+ GST_PTR_FORMAT ".", element, pad, caps);
+
+ /* Sanity checks */
+ if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
+ /* Sink pad, do not expose */
+ gst_caps_unref (caps);
+ return;
+ }
+
+ if (G_LIKELY (caps)) {
+ GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
+
+ if (gst_caps_can_intersect (caps, ref_caps)) {
+ /* SRC RTCP caps, do not expose */
+ gst_caps_unref (ref_caps);
+ gst_caps_unref (caps);
+
+ return;
+ }
+ gst_caps_unref (ref_caps);
+ } else {
+ GST_ERROR_OBJECT (self, "Pad with no caps detected.");
+ gst_caps_unref (caps);
+
+ return;
+ }
+ gst_caps_unref (caps);
+
+ GST_RTP_SRC_LOCK (self);
+ g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads);
+ upad = gst_ghost_pad_new (name, pad);
+
+ gst_pad_set_active (upad, TRUE);
+ gst_element_add_pad (GST_ELEMENT (self), upad);
+
+ GST_RTP_SRC_UNLOCK (self);
+}
+
+static void
+gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
+ gpointer data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (data);
+ GST_INFO_OBJECT (self,
+ "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
+ pad);
+}
+
+static void
+gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id,
+ guint ssrc, gpointer data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (data);
+
+ GST_INFO_OBJECT (self,
+ "Dectected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id,
+ ssrc);
+}
+
+static void
+gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id,
+ guint ssrc, gpointer data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (data);
+
+ GST_INFO_OBJECT (self, "Dectected a new SSRC: session-id 0x%x, ssrc 0x%x.",
+ session_id, ssrc);
+}
+
+static GstPadProbeReturn
+gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info,
+ gpointer user_data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (user_data);
+ GstBuffer *buffer;
+ GstNetAddressMeta *meta;
+
+ if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
+ GstBufferList *buffer_list = info->data;
+ buffer = gst_buffer_list_get (buffer_list, 0);
+ } else {
+ buffer = info->data;
+ }
+
+ meta = gst_buffer_get_net_address_meta (buffer);
+
+ GST_OBJECT_LOCK (self);
+ g_clear_object (&self->rtcp_send_addr);
+ self->rtcp_send_addr = g_object_ref (meta->addr);
+ GST_OBJECT_UNLOCK (self);
+
+ return GST_PAD_PROBE_OK;
+}
+
+static inline void
+gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer)
+{
+ GST_OBJECT_LOCK (self);
+ if (self->rtcp_send_addr)
+ gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr);
+ GST_OBJECT_UNLOCK (self);
+}
+
+static GstPadProbeReturn
+gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
+ gpointer user_data)
+{
+ GstRtpSrc *self = GST_RTP_SRC (user_data);
+
+ if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
+ GstBufferList *buffer_list = info->data;
+ GstBuffer *buffer;
+ gint i;
+
+ info->data = buffer_list = gst_buffer_list_make_writable (buffer_list);
+ for (i = 0; i < gst_buffer_list_length (buffer_list); i++) {
+ buffer = gst_buffer_list_get (buffer_list, i);
+ gst_rtp_src_attach_net_address_meta (self, buffer);
+ }
+ } else {
+ GstBuffer *buffer = info->data;
+ info->data = buffer = gst_buffer_make_writable (buffer);
+ gst_rtp_src_attach_net_address_meta (self, buffer);
+ }
+
+ return GST_PAD_PROBE_OK;
+}
+
+static gboolean
+gst_rtp_src_setup_elements (GstRtpSrc * self)
+{
+ GstPad *pad;
+ GSocket *socket;
+ GInetAddress *addr;
+ gchar name[48];
+ GstCaps *caps;
+ gchar *address;
+ guint rtcp_port;
+
+ /* Construct the RTP receiver pipeline.
+ *
+ * udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
+ * | rtpbin |
+ * udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
+ *
+ * This pipeline is fixed for now, note that optionally an FEC stream could
+ * be added later.
+ */
+
+ /* Should not be NULL */
+ g_return_val_if_fail (self->uri != NULL, FALSE);
+
+ self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
+ if (self->rtpbin == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtpbin element is not available"));
+ return FALSE;
+ }
+
+ self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
+ if (self->rtp_src == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtp_src element is not available"));
+ return FALSE;
+ }
+
+ self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
+ if (self->rtcp_src == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtcp_src element is not available"));
+ return FALSE;
+ }
+
+ self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
+ if (self->rtcp_sink == NULL) {
+ GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
+ ("%s", "rtcp_sink element is not available"));
+ return FALSE;
+ }
+
+ /* Add rtpbin callbacks to monitor the operation of rtpbin */
+ g_signal_connect (self->rtpbin, "pad-added",
+ G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
+ g_signal_connect (self->rtpbin, "pad-removed",
+ G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
+ g_signal_connect (self->rtpbin, "request-pt-map",
+ G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
+ g_signal_connect (self->rtpbin, "on-new-ssrc",
+ G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
+ g_signal_connect (self->rtpbin, "on-ssrc-collision",
+ G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
+
+ g_object_set (self->rtpbin, "latency", self->latency, NULL);
+
+ /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
+ * not all at the same moment */
+ gst_bin_add (GST_BIN (self), self->rtpbin);
+ gst_bin_add (GST_BIN (self), self->rtp_src);
+
+ g_object_set (self->rtp_src,
+ "address", gst_uri_get_host (self->uri),
+ "port", gst_uri_get_port (self->uri), NULL);
+
+ gst_bin_add (GST_BIN (self), self->rtcp_sink);
+
+ /* no need to set address if unicast */
+ caps = gst_caps_new_empty_simple ("application/x-rtcp");
+ g_object_set (self->rtcp_src,
+ "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
+ gst_caps_unref (caps);
+
+ addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
+ if (g_inet_address_get_is_multicast (addr)) {
+ g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
+ NULL);
+ }
+ g_object_unref (addr);
+
+ g_object_set (self->rtcp_sink,
+ "host", gst_uri_get_host (self->uri),
+ "port", gst_uri_get_port (self->uri) + 1,
+ "ttl", self->ttl, "ttl-mc", self->ttl_mc,
+ /* Set false since we're reusing a socket */
+ "auto-multicast", FALSE, NULL);
+
+ gst_bin_add (GST_BIN (self), self->rtcp_src);
+
+ /* share the socket created by the source */
+ g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket,
+ "address", &address, "port", &rtcp_port, NULL);
+
+ addr = g_inet_address_new_from_string (address);
+ g_free (address);
+
+ if (g_inet_address_get_is_multicast (addr)) {
+ /* mc-ttl is not supported by dynudpsink */
+ g_socket_set_multicast_ttl (socket, self->ttl_mc);
+ /* In multicast, send RTCP to the multicast group */
+ self->rtcp_send_addr = g_inet_socket_address_new (addr, rtcp_port);
+ } else {
+ /* In unicast, send RTCP to the detected sender address */
+ pad = gst_element_get_static_pad (self->rtcp_src, "src");
+ self->rtcp_recv_probe = gst_pad_add_probe (pad,
+ GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
+ gst_rtp_src_on_recv_rtcp, self, NULL);
+ gst_object_unref (pad);
+ }
+ g_object_unref (addr);
+
+ pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
+ self->rtcp_send_probe = gst_pad_add_probe (pad,
+ GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
+ gst_rtp_src_on_send_rtcp, self, NULL);
+ gst_object_unref (pad);
+
+ g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
+
+ /* pads are all named */
+ g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
+
+ g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
+
+ gst_element_sync_state_with_parent (self->rtpbin);
+ gst_element_sync_state_with_parent (self->rtp_src);
+ gst_element_sync_state_with_parent (self->rtcp_sink);
+
+ g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
+ gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
+
+ gst_element_sync_state_with_parent (self->rtcp_src);
+
+ return TRUE;
+}
+
+static void
+gst_rtp_src_stop (GstRtpSrc * self)
+{
+ GstPad *pad;
+
+ if (self->rtcp_recv_probe) {
+ pad = gst_element_get_static_pad (self->rtcp_src, "src");
+ gst_pad_remove_probe (pad, self->rtcp_recv_probe);
+ self->rtcp_recv_probe = 0;
+ gst_object_unref (pad);
+ }
+
+ pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
+ gst_pad_remove_probe (pad, self->rtcp_send_probe);
+ self->rtcp_send_probe = 0;
+ gst_object_unref (pad);
+}
+
+static GstStateChangeReturn
+gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
+{
+ GstRtpSrc *self = GST_RTP_SRC (element);
+ GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+ GST_DEBUG_OBJECT (self, "Changing state: %s => %s",
+ gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
+ gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ if (gst_rtp_src_setup_elements (self) == FALSE)
+ return GST_STATE_CHANGE_FAILURE;
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ return ret;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ ret = GST_STATE_CHANGE_NO_PREROLL;
+ break;
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ ret = GST_STATE_CHANGE_NO_PREROLL;
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ gst_rtp_src_stop (self);
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+static void
+gst_rtp_src_init (GstRtpSrc * self)
+{
+ self->rtpbin = NULL;
+ self->rtp_src = NULL;
+ self->rtcp_src = NULL;
+ self->rtcp_sink = NULL;
+
+ self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
+ self->ttl = DEFAULT_PROP_TTL;
+ self->ttl_mc = DEFAULT_PROP_TTL_MC;
+ self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
+ self->latency = DEFAULT_PROP_LATENCY;
+
+ GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
+ gst_bin_set_suppressed_flags (GST_BIN (self),
+ GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
+
+ g_mutex_init (&self->lock);
+}
+
+static guint
+gst_rtp_src_uri_get_type (GType type)
+{
+ return GST_URI_SRC;
+}
+
+static const gchar *const *
+gst_rtp_src_uri_get_protocols (GType type)
+{
+ static const gchar *protocols[] = { (char *) "rtp", NULL };
+
+ return protocols;
+}
+
+static gchar *
+gst_rtp_src_uri_get_uri (GstURIHandler * handler)
+{
+ GstRtpSrc *self = (GstRtpSrc *) handler;
+
+ return gst_uri_to_string (self->uri);
+}
+
+static gboolean
+gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+ GError ** error)
+{
+ GstRtpSrc *self = (GstRtpSrc *) handler;
+
+ g_object_set (G_OBJECT (self), "uri", uri, NULL);
+
+ return TRUE;
+}
+
+static void
+gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+ GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+ iface->get_type = gst_rtp_src_uri_get_type;
+ iface->get_protocols = gst_rtp_src_uri_get_protocols;
+ iface->get_uri = gst_rtp_src_uri_get_uri;
+ iface->set_uri = gst_rtp_src_uri_set_uri;
+}
+
+/* ex: set tabstop=2 shiftwidth=2 expandtab: */
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2019 Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTP_SRC_H__
+#define __GST_RTP_SRC_H__
+
+#include <gio/gio.h>
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_RTP_SRC \
+ (gst_rtp_src_get_type())
+#define GST_RTP_SRC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTP_SRC, GstRtpSrc))
+#define GST_RTP_SRC_CAST(obj) \
+ ((GstRtpSrc *) obj)
+#define GST_RTP_SRC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTP_SRC, GstRtpSrcClass))
+#define GST_IS_RTP_SRC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTP_SRC))
+#define GST_IS_RTP_SRC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTP_SRC))
+
+typedef struct _GstRtpSrc GstRtpSrc;
+typedef struct _GstRtpSrcClass GstRtpSrcClass;
+
+struct _GstRtpSrc
+{
+ GstBin parent;
+
+ /* Properties */
+ GstUri *uri;
+ gint ttl;
+ gint ttl_mc;
+ gint latency;
+ gchar *encoding_name;
+ guint latency_ms;
+
+ /* Internal elements */
+ GstElement *rtpbin;
+ GstElement *rtp_src;
+ GstElement *rtcp_src;
+ GstElement *rtcp_sink;
+
+ gulong rtcp_recv_probe;
+ gulong rtcp_send_probe;
+ GSocketAddress *rtcp_send_addr;
+
+ GMutex lock;
+};
+
+struct _GstRtpSrcClass
+{
+ GstBinClass parent;
+};
+
+GType gst_rtp_src_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_RTP_SRC_H__ */
--- /dev/null
+gst_plugins_rtp_sources = [
+ 'plugin.c',
+ 'gstrtpsink.c',
+ 'gstrtpsrc.c',
+ 'gstrtp-utils.c',
+]
+
+gstrtp = library('gstrtpmanagerbad',
+ gst_plugins_rtp_sources,
+ dependencies: [gio_dep, gst_dep, gstbase_dep, gstrtp_dep, gstnet_dep, gstcontroller_dep],
+ include_directories: [configinc],
+ install: true,
+ c_args: gst_plugins_bad_args,
+ install_dir: plugins_install_dir,
+)
--- /dev/null
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstrtpsink.h"
+#include "gstrtpsrc.h"
+
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+
+ gboolean ret = FALSE;
+
+ ret |= gst_element_register (plugin, "rtpsrc",
+ GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SRC);
+
+ ret |= gst_element_register (plugin, "rtpsink",
+ GST_RANK_PRIMARY + 1, GST_TYPE_RTP_SINK);
+
+ return ret;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ rtpmanagerbad,
+ "GStreamer RTP Plugins",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
option('rawparse', type : 'feature', value : 'auto')
option('removesilence', type : 'feature', value : 'auto')
option('rist', type : 'feature', value : 'auto')
+option('rtp', type : 'feature', value : 'auto')
option('sdp', type : 'feature', value : 'auto')
option('segmentclip', type : 'feature', value : 'auto')
option('siren', type : 'feature', value : 'auto')
elements/pnm \
elements/rtponvifparse \
elements/rtponviftimestamp \
+ elements/rtpsrc \
+ elements/rtpsink \
elements/id3mux \
pipelines/mxf \
libs/isoff \
elements_rtponviftimestamp_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_rtponviftimestamp_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+elements_rtpsrc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
+elements_rtpsrc_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
+elements_rtpsink_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
+elements_rtpsink_LDADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
EXTRA_DIST = gst-plugins-bad.supp $(uvch264_dist_data)
orc_bayer_CFLAGS = $(ORC_CFLAGS)
pnm
rtponvifparse
rtponviftimestamp
+rtpsrc
+rtpsink
shm
srtp
templatematch
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+GST_START_TEST (test_uri_to_properties)
+{
+ GstElement *rtpsink;
+
+ gint ttl, ttl_mc;
+
+ rtpsink = gst_element_factory_make ("rtpsink", NULL);
+
+ /* Sets properties to non-default values (make sure this stays in sync) */
+ g_object_set (rtpsink, "uri", "rtp://1.230.1.2?" "ttl=8" "&ttl-mc=9", NULL);
+
+ g_object_get (rtpsink, "ttl", &ttl, "ttl_mc", &ttl_mc, NULL);
+
+ /* Make sure these values are in sync with the one from the URI. */
+ g_assert_cmpint (ttl, ==, 8);
+ g_assert_cmpint (ttl_mc, ==, 9);
+
+ gst_object_unref (rtpsink);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpsink_suite (void)
+{
+ Suite *s = suite_create ("rtpsink");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_uri_to_properties);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtpsink);
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+
+GST_START_TEST (test_uri_to_properties)
+{
+ GstElement *rtpsrc;
+ guint latency, ttl, ttl_mc;
+
+ rtpsrc = gst_element_factory_make ("rtpsrc", NULL);
+
+ /* Sets properties to non-default values (make sure this stays in sync) */
+ g_object_set (rtpsrc, "uri", "rtp://1.230.1.2?"
+ "latency=300" "&ttl=8" "&ttl-mc=9", NULL);
+
+ g_object_get (rtpsrc,
+ "latency", &latency, "ttl-mc", &ttl_mc, "ttl", &ttl, NULL);
+
+ /* Make sure these values are in sync with the one from the URI. */
+ g_assert_cmpuint (latency, ==, 300);
+ g_assert_cmpint (ttl, ==, 8);
+ g_assert_cmpint (ttl_mc, ==, 9);
+
+ gst_object_unref (rtpsrc);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpsrc_suite (void)
+{
+ Suite *s = suite_create ("rtpsrc");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_uri_to_properties);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtpsrc);
[['elements/pnm.c']],
[['elements/rtponvifparse.c']],
[['elements/rtponviftimestamp.c']],
+ [['elements/rtpsrc.c']],
+ [['elements/rtpsink.c']],
[['elements/videoframe-audiolevel.c']],
[['elements/viewfinderbin.c']],
[['libs/h264parser.c'], false, [gstcodecparsers_dep]],
projects / platform / upstream / gstreamer.git / commitdiff